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# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
group("api") {
deps = [
":libjingle_peerconnection",
]
}
rtc_source_set("call_api") {
sources = [
"call/audio_receive_stream.h",
"call/audio_send_stream.h",
"call/audio_sink.h",
"call/audio_state.h",
]
deps = [
# TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
"..:webrtc_common",
"../base:rtc_base_approved",
"../modules/audio_coding:audio_encoder_interface",
]
}
config("libjingle_peerconnection_warnings_config") {
# GN orders flags on a target before flags from configs. The default config
# adds these flags so to cancel them out they need to come from a config and
# cannot be on the target directly.
if (!is_win && !is_clang) {
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
}
}
rtc_source_set("libjingle_peerconnection") {
cflags = []
sources = [
"audiotrack.cc",
"audiotrack.h",
"datachannel.cc",
"datachannel.h",
"datachannelinterface.h",
"dtmfsender.cc",
"dtmfsender.h",
"dtmfsenderinterface.h",
"jsep.h",
"jsepicecandidate.cc",
"jsepicecandidate.h",
"jsepsessiondescription.cc",
"jsepsessiondescription.h",
"localaudiosource.cc",
"localaudiosource.h",
"mediaconstraintsinterface.cc",
"mediaconstraintsinterface.h",
"mediacontroller.cc",
"mediacontroller.h",
"mediastream.cc",
"mediastream.h",
"mediastreaminterface.h",
"mediastreamobserver.cc",
"mediastreamobserver.h",
"mediastreamproxy.h",
"mediastreamtrack.h",
"mediastreamtrackproxy.h",
"notifier.h",
"peerconnection.cc",
"peerconnection.h",
"peerconnectionfactory.cc",
"peerconnectionfactory.h",
"peerconnectionfactoryproxy.h",
"peerconnectioninterface.h",
"peerconnectionproxy.h",
"proxy.h",
"remoteaudiosource.cc",
"remoteaudiosource.h",
RTCStats and RTCStatsReport added (webrtc/stats). The old and new getStats are very different. This CL proposes rewriting the new getStats from scratch with a bottom-up approach, starting with the fundamental stats classes. This will allow cleaner and more efficient code that is more aligned with the spec. RTCStats and subclasses are the equivalent to RTCStats and RTCStats- -derived dictionaries from the specs[1][2]. The dictionary members are public member variables of type RTCStatsMember<T>, where T is one of the supported types. All members derive from RTCStatsMemberInterface and iteration of members is possible with RTCStats::Members(). The members are not stored in a map for performance and readability. Type checking is supported with static class variables, kType. Only the supported member types T are specialized and may be instantiated, and sequences are supported with std::vector<...>. Type checking is again supported with static class variables, kType. RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id to RTCStats-objects. RTCStatsReport is reference counted. It and its contained stats may be destroyed on any thread. When the RTCStatsCollector is added in a follow-up CL, it will return const references to the RTCStatsReports. This means copies don't have to be made for multiple stats observers or when jumping threads. In fact, no copies of any stats will have to be made in surfacing stats to Blink. [1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary [2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html [3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object This adds the new folder webrtc/stats/, with target rtc_stats and binary rtc_stats_unittests. Public api headers are placed in webrtc/api/ and .cc files are placed in webrtc/stats/. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2241093002 Cr-Commit-Position: refs/heads/master@{#13879}
2016-08-24 01:33:13 -07:00
"rtcstats.h",
"rtcstats_objects.h",
RTCStats and RTCStatsReport added (webrtc/stats). The old and new getStats are very different. This CL proposes rewriting the new getStats from scratch with a bottom-up approach, starting with the fundamental stats classes. This will allow cleaner and more efficient code that is more aligned with the spec. RTCStats and subclasses are the equivalent to RTCStats and RTCStats- -derived dictionaries from the specs[1][2]. The dictionary members are public member variables of type RTCStatsMember<T>, where T is one of the supported types. All members derive from RTCStatsMemberInterface and iteration of members is possible with RTCStats::Members(). The members are not stored in a map for performance and readability. Type checking is supported with static class variables, kType. Only the supported member types T are specialized and may be instantiated, and sequences are supported with std::vector<...>. Type checking is again supported with static class variables, kType. RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id to RTCStats-objects. RTCStatsReport is reference counted. It and its contained stats may be destroyed on any thread. When the RTCStatsCollector is added in a follow-up CL, it will return const references to the RTCStatsReports. This means copies don't have to be made for multiple stats observers or when jumping threads. In fact, no copies of any stats will have to be made in surfacing stats to Blink. [1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary [2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html [3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object This adds the new folder webrtc/stats/, with target rtc_stats and binary rtc_stats_unittests. Public api headers are placed in webrtc/api/ and .cc files are placed in webrtc/stats/. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2241093002 Cr-Commit-Position: refs/heads/master@{#13879}
2016-08-24 01:33:13 -07:00
"rtcstatsreport.h",
"rtpparameters.h",
"rtpreceiver.cc",
"rtpreceiver.h",
"rtpreceiverinterface.h",
"rtpsender.cc",
"rtpsender.h",
"rtpsenderinterface.h",
"sctputils.cc",
"sctputils.h",
"statscollector.cc",
"statscollector.h",
"statstypes.cc",
"statstypes.h",
"streamcollection.h",
"videocapturertracksource.cc",
"videocapturertracksource.h",
"videosourceproxy.h",
"videotrack.cc",
"videotrack.h",
"videotracksource.cc",
"videotracksource.h",
"webrtcsdp.cc",
"webrtcsdp.h",
"webrtcsession.cc",
"webrtcsession.h",
"webrtcsessiondescriptionfactory.cc",
"webrtcsessiondescriptionfactory.h",
]
configs += [ ":libjingle_peerconnection_warnings_config" ]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":call_api",
"../call",
"../media",
"../pc",
]
if (rtc_use_quic) {
sources += [
"quicdatachannel.cc",
"quicdatachannel.h",
"quicdatatransport.cc",
"quicdatatransport.h",
]
deps += [ "//third_party/libquic" ]
public_deps = [
"//third_party/libquic",
]
}
}
# Exclude the targets below from the Chromium build since they cannot be built
# due to incompability with Chromium's logging implementation.
if (is_android && !build_with_chromium) {
config("libjingle_peerconnection_jni_warnings_config") {
# The warnings below are enabled by default. Since GN orders compiler flags
# for a target before flags from configs, the only way to disable such
# warnings is by having them in a separate config, loaded from the target.
if (!is_win) {
cflags = [
"-Wno-sign-compare",
"-Wno-unused-variable",
]
}
}
rtc_source_set("libjingle_peerconnection_jni") {
sources = [
"android/jni/androidmediacodeccommon.h",
"android/jni/androidmediadecoder_jni.cc",
"android/jni/androidmediadecoder_jni.h",
"android/jni/androidmediaencoder_jni.cc",
"android/jni/androidmediaencoder_jni.h",
"android/jni/androidmetrics_jni.cc",
"android/jni/androidnetworkmonitor_jni.cc",
"android/jni/androidnetworkmonitor_jni.h",
"android/jni/androidvideotracksource_jni.cc",
"android/jni/classreferenceholder.cc",
"android/jni/classreferenceholder.h",
"android/jni/jni_helpers.cc",
"android/jni/jni_helpers.h",
"android/jni/native_handle_impl.cc",
"android/jni/native_handle_impl.h",
"android/jni/peerconnection_jni.cc",
"android/jni/surfacetexturehelper_jni.cc",
"android/jni/surfacetexturehelper_jni.h",
"androidvideotracksource.cc",
"androidvideotracksource.h",
]
configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [
"//build/config/clang:extra_warnings",
"//build/config/clang:find_bad_constructs",
]
}
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (is_win) {
cflags += [
"/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
"/wd4389", # signed/unsigned mismatch.
]
}
deps = [
":libjingle_peerconnection",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
]
if (rtc_build_libyuv) {
deps += [ "$rtc_libyuv_dir" ]
public_deps = [
"$rtc_libyuv_dir",
]
} else {
# Need to add a directory normally exported by libyuv.
include_dirs += [ "$rtc_libyuv_dir/include" ]
}
}
rtc_shared_library("libjingle_peerconnection_so") {
sources = [
"android/jni/jni_onload.cc",
]
suppressed_configs += [ "//build/config/android:hide_native_jni_exports" ]
deps = [
":libjingle_peerconnection",
":libjingle_peerconnection_jni",
]
output_extension = "so"
}
android_library("libjingle_peerconnection_java") {
java_files = [
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
"../modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
"android/java/src/org/webrtc/AudioSource.java",
"android/java/src/org/webrtc/AudioTrack.java",
"android/java/src/org/webrtc/CallSessionFileRotatingLogSink.java",
"android/java/src/org/webrtc/Camera1Enumerator.java",
"android/java/src/org/webrtc/Camera2Capturer.java",
"android/java/src/org/webrtc/Camera2Enumerator.java",
"android/java/src/org/webrtc/Camera2Session.java",
"android/java/src/org/webrtc/CameraCapturer.java",
"android/java/src/org/webrtc/CameraEnumerationAndroid.java",
"android/java/src/org/webrtc/CameraEnumerator.java",
"android/java/src/org/webrtc/CameraSession.java",
"android/java/src/org/webrtc/CameraVideoCapturer.java",
"android/java/src/org/webrtc/DataChannel.java",
"android/java/src/org/webrtc/EglBase.java",
"android/java/src/org/webrtc/EglBase10.java",
"android/java/src/org/webrtc/EglBase14.java",
"android/java/src/org/webrtc/GlRectDrawer.java",
"android/java/src/org/webrtc/GlShader.java",
"android/java/src/org/webrtc/GlTextureFrameBuffer.java",
"android/java/src/org/webrtc/GlUtil.java",
"android/java/src/org/webrtc/IceCandidate.java",
"android/java/src/org/webrtc/MediaCodecVideoDecoder.java",
"android/java/src/org/webrtc/MediaCodecVideoEncoder.java",
"android/java/src/org/webrtc/MediaConstraints.java",
"android/java/src/org/webrtc/MediaSource.java",
"android/java/src/org/webrtc/MediaStream.java",
"android/java/src/org/webrtc/MediaStreamTrack.java",
"android/java/src/org/webrtc/Metrics.java",
"android/java/src/org/webrtc/NetworkMonitor.java",
"android/java/src/org/webrtc/NetworkMonitorAutoDetect.java",
"android/java/src/org/webrtc/PeerConnection.java",
"android/java/src/org/webrtc/PeerConnectionFactory.java",
"android/java/src/org/webrtc/RendererCommon.java",
"android/java/src/org/webrtc/RtpParameters.java",
"android/java/src/org/webrtc/RtpReceiver.java",
"android/java/src/org/webrtc/RtpSender.java",
"android/java/src/org/webrtc/SdpObserver.java",
"android/java/src/org/webrtc/SessionDescription.java",
"android/java/src/org/webrtc/StatsObserver.java",
"android/java/src/org/webrtc/StatsReport.java",
"android/java/src/org/webrtc/SurfaceTextureHelper.java",
"android/java/src/org/webrtc/SurfaceViewRenderer.java",
"android/java/src/org/webrtc/VideoCapturer.java",
"android/java/src/org/webrtc/VideoCapturerAndroid.java",
"android/java/src/org/webrtc/VideoRenderer.java",
"android/java/src/org/webrtc/VideoRendererGui.java",
"android/java/src/org/webrtc/VideoSource.java",
"android/java/src/org/webrtc/VideoTrack.java",
]
deps = [
"//webrtc/base:base_java",
]
}
}
if (rtc_include_tests) {
config("peerconnection_unittests_config") {
# The warnings below are enabled by default. Since GN orders compiler flags
# for a target before flags from configs, the only way to disable such
# warnings is by having them in a separate config, loaded from the target.
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.webrtc.org/3307.
if (is_clang && is_win) {
cflags = [
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
# for -Wno-sign-compare
"-Wno-sign-compare",
"-Wno-unused-function",
]
}
if (!is_win) {
cflags = [ "-Wno-sign-compare" ]
cflags_cc = [ "-Wno-overloaded-virtual" ]
}
}
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
"datachannel_unittest.cc",
"dtmfsender_unittest.cc",
"fakemetricsobserver.cc",
"fakemetricsobserver.h",
"jsepsessiondescription_unittest.cc",
"localaudiosource_unittest.cc",
"mediaconstraintsinterface_unittest.cc",
"mediastream_unittest.cc",
"peerconnection_unittest.cc",
"peerconnectionendtoend_unittest.cc",
"peerconnectionfactory_unittest.cc",
"peerconnectioninterface_unittest.cc",
"proxy_unittest.cc",
"rtpsenderreceiver_unittest.cc",
"statscollector_unittest.cc",
"test/fakeaudiocapturemodule.cc",
"test/fakeaudiocapturemodule.h",
"test/fakeaudiocapturemodule_unittest.cc",
"test/fakeconstraints.h",
"test/fakedatachannelprovider.h",
"test/fakeperiodicvideocapturer.h",
"test/fakertccertificategenerator.h",
"test/fakevideotrackrenderer.h",
"test/mock_datachannel.h",
"test/mock_peerconnection.h",
"test/mock_webrtcsession.h",
"test/mockpeerconnectionobservers.h",
"test/peerconnectiontestwrapper.cc",
"test/peerconnectiontestwrapper.h",
"test/testsdpstrings.h",
"videocapturertracksource_unittest.cc",
"videotrack_unittest.cc",
"webrtcsdp_unittest.cc",
"webrtcsession_unittest.cc",
]
defines = [ "HAVE_SCTP" ]
configs += [ ":peerconnection_unittests_config" ]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (is_win) {
cflags = [
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
"/wd4389", # signed/unsigned mismatch.
]
}
if (rtc_use_quic) {
public_deps = [
"//third_party/libquic",
]
sources += [
"quicdatachannel_unittest.cc",
"quicdatatransport_unittest.cc",
]
}
deps = []
if (is_android) {
sources += [
"test/androidtestinitializer.cc",
"test/androidtestinitializer.h",
]
deps += [
":libjingle_peerconnection_java",
":libjingle_peerconnection_jni",
"//testing/android/native_test:native_test_support",
]
}
deps += [
":libjingle_peerconnection",
"..:webrtc_common",
"../base:rtc_base_tests_utils",
"../media:rtc_unittest_main",
"../pc:rtc_pc",
"../system_wrappers:metrics_default",
"//testing/gmock",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
}
if (is_android) {
instrumentation_test_apk("libjingle_peerconnection_android_unittest") {
apk_name = "libjingle_peerconnection_android_unittest"
android_manifest = "androidtests/AndroidManifest.xml"
java_files = [
"androidtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java",
"androidtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java",
"androidtests/src/org/webrtc/Camera2CapturerTest.java",
"androidtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java",
"androidtests/src/org/webrtc/GlRectDrawerTest.java",
"androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java",
"androidtests/src/org/webrtc/NetworkMonitorTest.java",
"androidtests/src/org/webrtc/PeerConnectionTest.java",
"androidtests/src/org/webrtc/RendererCommonTest.java",
"androidtests/src/org/webrtc/SurfaceTextureHelperTest.java",
"androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java",
]
deps = [
":libjingle_peerconnection_android_unittest_resources",
":libjingle_peerconnection_java",
"//base:base_java",
"//webrtc/base:base_java",
]
shared_libraries = [ ":libjingle_peerconnection_so" ]
}
android_resources("libjingle_peerconnection_android_unittest_resources") {
resource_dirs = [ "androidtests/res" ]
custom_package = "org.webrtc"
}
}
}